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diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.cc
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-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h"
-
-#include <assert.h>
-#include <string.h> // Access to memset.
-
-#include <algorithm> // Access to min, max.
-
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-
-namespace webrtc {
-
-// Table of constants used in method DspHelper::ParabolicFit().
-const int16_t DspHelper::kParabolaCoefficients[17][3] = {
- { 120, 32, 64 },
- { 140, 44, 75 },
- { 150, 50, 80 },
- { 160, 57, 85 },
- { 180, 72, 96 },
- { 200, 89, 107 },
- { 210, 98, 112 },
- { 220, 108, 117 },
- { 240, 128, 128 },
- { 260, 150, 139 },
- { 270, 162, 144 },
- { 280, 174, 149 },
- { 300, 200, 160 },
- { 320, 228, 171 },
- { 330, 242, 176 },
- { 340, 257, 181 },
- { 360, 288, 192 } };
-
-// Filter coefficients used when downsampling from the indicated sample rates
-// (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. The corresponding Q0
-// values are provided in the comments before each array.
-
-// Q0 values: {0.3, 0.4, 0.3}.
-const int16_t DspHelper::kDownsample8kHzTbl[3] = { 1229, 1638, 1229 };
-
-// Q0 values: {0.15, 0.2, 0.3, 0.2, 0.15}.
-const int16_t DspHelper::kDownsample16kHzTbl[5] = { 614, 819, 1229, 819, 614 };
-
-// Q0 values: {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425}.
-const int16_t DspHelper::kDownsample32kHzTbl[7] = {
- 584, 512, 625, 667, 625, 512, 584 };
-
-// Q0 values: {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487}.
-const int16_t DspHelper::kDownsample48kHzTbl[7] = {
- 1019, 390, 427, 440, 427, 390, 1019 };
-
-int DspHelper::RampSignal(const int16_t* input,
- size_t length,
- int factor,
- int increment,
- int16_t* output) {
- int factor_q20 = (factor << 6) + 32;
- // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
- for (size_t i = 0; i < length; ++i) {
- output[i] = (factor * input[i] + 8192) >> 14;
- factor_q20 += increment;
- factor_q20 = std::max(factor_q20, 0); // Never go negative.
- factor = std::min(factor_q20 >> 6, 16384);
- }
- return factor;
-}
-
-int DspHelper::RampSignal(int16_t* signal,
- size_t length,
- int factor,
- int increment) {
- return RampSignal(signal, length, factor, increment, signal);
-}
-
-int DspHelper::RampSignal(AudioMultiVector* signal,
- size_t start_index,
- size_t length,
- int factor,
- int increment) {
- assert(start_index + length <= signal->Size());
- if (start_index + length > signal->Size()) {
- // Wrong parameters. Do nothing and return the scale factor unaltered.
- return factor;
- }
- int end_factor = 0;
- // Loop over the channels, starting at the same |factor| each time.
- for (size_t channel = 0; channel < signal->Channels(); ++channel) {
- end_factor =
- RampSignal(&(*signal)[channel][start_index], length, factor, increment);
- }
- return end_factor;
-}
-
-void DspHelper::PeakDetection(int16_t* data, int data_length,
- int num_peaks, int fs_mult,
- int* peak_index, int16_t* peak_value) {
- int16_t min_index = 0;
- int16_t max_index = 0;
-
- for (int i = 0; i <= num_peaks - 1; i++) {
- if (num_peaks == 1) {
- // Single peak. The parabola fit assumes that an extra point is
- // available; worst case it gets a zero on the high end of the signal.
- // TODO(hlundin): This can potentially get much worse. It breaks the
- // API contract, that the length of |data| is |data_length|.
- data_length++;
- }
-
- peak_index[i] = WebRtcSpl_MaxIndexW16(data, data_length - 1);
-
- if (i != num_peaks - 1) {
- min_index = std::max(0, peak_index[i] - 2);
- max_index = std::min(data_length - 1, peak_index[i] + 2);
- }
-
- if ((peak_index[i] != 0) && (peak_index[i] != (data_length - 2))) {
- ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
- &peak_value[i]);
- } else {
- if (peak_index[i] == data_length - 2) {
- if (data[peak_index[i]] > data[peak_index[i] + 1]) {
- ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
- &peak_value[i]);
- } else if (data[peak_index[i]] <= data[peak_index[i] + 1]) {
- // Linear approximation.
- peak_value[i] = (data[peak_index[i]] + data[peak_index[i] + 1]) >> 1;
- peak_index[i] = (peak_index[i] * 2 + 1) * fs_mult;
- }
- } else {
- peak_value[i] = data[peak_index[i]];
- peak_index[i] = peak_index[i] * 2 * fs_mult;
- }
- }
-
- if (i != num_peaks - 1) {
- memset(&data[min_index], 0,
- sizeof(data[0]) * (max_index - min_index + 1));
- }
- }
-}
-
-void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
- int* peak_index, int16_t* peak_value) {
- uint16_t fit_index[13];
- if (fs_mult == 1) {
- fit_index[0] = 0;
- fit_index[1] = 8;
- fit_index[2] = 16;
- } else if (fs_mult == 2) {
- fit_index[0] = 0;
- fit_index[1] = 4;
- fit_index[2] = 8;
- fit_index[3] = 12;
- fit_index[4] = 16;
- } else if (fs_mult == 4) {
- fit_index[0] = 0;
- fit_index[1] = 2;
- fit_index[2] = 4;
- fit_index[3] = 6;
- fit_index[4] = 8;
- fit_index[5] = 10;
- fit_index[6] = 12;
- fit_index[7] = 14;
- fit_index[8] = 16;
- } else {
- fit_index[0] = 0;
- fit_index[1] = 1;
- fit_index[2] = 3;
- fit_index[3] = 4;
- fit_index[4] = 5;
- fit_index[5] = 7;
- fit_index[6] = 8;
- fit_index[7] = 9;
- fit_index[8] = 11;
- fit_index[9] = 12;
- fit_index[10] = 13;
- fit_index[11] = 15;
- fit_index[12] = 16;
- }
-
- // num = -3 * signal_points[0] + 4 * signal_points[1] - signal_points[2];
- // den = signal_points[0] - 2 * signal_points[1] + signal_points[2];
- int32_t num = (signal_points[0] * -3) + (signal_points[1] * 4)
- - signal_points[2];
- int32_t den = signal_points[0] + (signal_points[1] * -2) + signal_points[2];
- int32_t temp = num * 120;
- int flag = 1;
- int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0]
- - kParabolaCoefficients[fit_index[fs_mult - 1]][0];
- int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0]
- + kParabolaCoefficients[fit_index[fs_mult - 1]][0]) / 2;
- int16_t lmt;
- if (temp < -den * strt) {
- lmt = strt - stp;
- while (flag) {
- if ((flag == fs_mult) || (temp > -den * lmt)) {
- *peak_value = (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1]
- + num * kParabolaCoefficients[fit_index[fs_mult - flag]][2]
- + signal_points[0] * 256) / 256;
- *peak_index = *peak_index * 2 * fs_mult - flag;
- flag = 0;
- } else {
- flag++;
- lmt -= stp;
- }
- }
- } else if (temp > -den * (strt + stp)) {
- lmt = strt + 2 * stp;
- while (flag) {
- if ((flag == fs_mult) || (temp < -den * lmt)) {
- int32_t temp_term_1 =
- den * kParabolaCoefficients[fit_index[fs_mult+flag]][1];
- int32_t temp_term_2 =
- num * kParabolaCoefficients[fit_index[fs_mult+flag]][2];
- int32_t temp_term_3 = signal_points[0] * 256;
- *peak_value = (temp_term_1 + temp_term_2 + temp_term_3) / 256;
- *peak_index = *peak_index * 2 * fs_mult + flag;
- flag = 0;
- } else {
- flag++;
- lmt += stp;
- }
- }
- } else {
- *peak_value = signal_points[1];
- *peak_index = *peak_index * 2 * fs_mult;
- }
-}
-
-int DspHelper::MinDistortion(const int16_t* signal, int min_lag,
- int max_lag, int length,
- int32_t* distortion_value) {
- int best_index = -1;
- int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
- for (int i = min_lag; i <= max_lag; i++) {
- int32_t sum_diff = 0;
- const int16_t* data1 = signal;
- const int16_t* data2 = signal - i;
- for (int j = 0; j < length; j++) {
- sum_diff += WEBRTC_SPL_ABS_W32(data1[j] - data2[j]);
- }
- // Compare with previous minimum.
- if (sum_diff < min_distortion) {
- min_distortion = sum_diff;
- best_index = i;
- }
- }
- *distortion_value = min_distortion;
- return best_index;
-}
-
-void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2,
- size_t length, int16_t* mix_factor,
- int16_t factor_decrement, int16_t* output) {
- int16_t factor = *mix_factor;
- int16_t complement_factor = 16384 - factor;
- for (size_t i = 0; i < length; i++) {
- output[i] =
- (factor * input1[i] + complement_factor * input2[i] + 8192) >> 14;
- factor -= factor_decrement;
- complement_factor += factor_decrement;
- }
- *mix_factor = factor;
-}
-
-void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
- int16_t* factor, int16_t increment,
- int16_t* output) {
- uint16_t factor_16b = *factor;
- int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
- for (size_t i = 0; i < length; i++) {
- output[i] = (factor_16b * input[i] + 8192) >> 14;
- factor_32b = std::max(factor_32b + increment, 0);
- factor_16b = std::min(16384, factor_32b >> 6);
- }
- *factor = factor_16b;
-}
-
-void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
- int32_t factor = (16384 << 6) + 32;
- for (size_t i = 0; i < length; i++) {
- signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
- factor -= mute_slope;
- }
-}
-
-int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
- int output_length, int input_rate_hz,
- bool compensate_delay, int16_t* output) {
- // Set filter parameters depending on input frequency.
- // NOTE: The phase delay values are wrong compared to the true phase delay
- // of the filters. However, the error is preserved (through the +1 term) for
- // consistency.
- const int16_t* filter_coefficients; // Filter coefficients.
- int16_t filter_length; // Number of coefficients.
- int16_t filter_delay; // Phase delay in samples.
- int16_t factor; // Conversion rate (inFsHz / 8000).
- switch (input_rate_hz) {
- case 8000: {
- filter_length = 3;
- factor = 2;
- filter_coefficients = kDownsample8kHzTbl;
- filter_delay = 1 + 1;
- break;
- }
- case 16000: {
- filter_length = 5;
- factor = 4;
- filter_coefficients = kDownsample16kHzTbl;
- filter_delay = 2 + 1;
- break;
- }
- case 32000: {
- filter_length = 7;
- factor = 8;
- filter_coefficients = kDownsample32kHzTbl;
- filter_delay = 3 + 1;
- break;
- }
- case 48000: {
- filter_length = 7;
- factor = 12;
- filter_coefficients = kDownsample48kHzTbl;
- filter_delay = 3 + 1;
- break;
- }
- default: {
- assert(false);
- return -1;
- }
- }
-
- if (!compensate_delay) {
- // Disregard delay compensation.
- filter_delay = 0;
- }
-
- // Returns -1 if input signal is too short; 0 otherwise.
- return WebRtcSpl_DownsampleFast(
- &input[filter_length - 1], static_cast<int>(input_length) -
- (filter_length - 1), output, output_length, filter_coefficients,
- filter_length, factor, filter_delay);
-}
-
-} // namespace webrtc
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